Grandstream Networks, Inc.
GXP – 2000
SIP Enterprise Phone
GXP– 2000 User’s Manual
Firmware Version 1.1.1.14
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TABLE OF FIGURES
GXW- 2000 USER’S MANUAL
FIGURE 1: CONNECTING THE GXP-2000 ......................................................................................................... 5
FIGURE 2: CONNECTING THE GXP-2000 AND THE GXP-EXTENSION................................................................. 7
FIGURE 3: GXP-2000 INTERNAL HEADSET WIRING SCHEMA ............................................................................ 7
FIGURE 4: GETTING FAMILIAR WITH KEYPAD .................................................................................................. 11
TABLE OF TABLES
GXW- 2000 USER’S MANUAL
TABLE 1: DEFINITIONS OF THE GXP- 2000 CONNECTORS................................................................................. 5
TABLE 2: GXP KEY FEATURES IN A GLANCE .................................................................................................... 8
TABLE 3: HARDWARE SPECIFICATIONS ............................................................................................................ 8
TABLE 4: GXP-2000 TECHNICAL SPECIFICATIONS ........................................................................................... 9
TABLE 5: LCD ICON DEFINITIONS .................................................................................................................. 10
TABLE 7: GXP-2000 CALL FEATURES ........................................................................................................... 15
TABLE 8: KEY PAD CONFIGURATION MENU .................................................................................................... 16
TABLE 9: DEVICE CONFIGURATION – BASIC SETTINGS DEFINITIONS ................................................................ 18
TABLE 10: DEVICE CONFIGURATION - STATUS DEFINITIONS............................................................................ 19
TABLE 11: ADVANCED USER CONFIGURATION PAGE DEFINITIONS................................................................... 20
TABLE 12: SIP ACCOUNT CONFIGURATION PAGE DEFINITIONS ....................................................................... 23
GUI INTERFACE EXAMPLES
GXW- 2000 USER’S MANUAL
1. SCREENSHOT OF CONFIGURATION LOGIN PAGE
2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE
3. SCREENSHOT OF STATUS CONFIGURATION PAGE
4. STATUS CONFIGURATION PAGE DEFINITIONS
5. SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE
6. SCREENSHOT OF SIP ACCOUNT CONFIGURATION
7. SCREENSHOT OF SAVED CONFIGURATION CHANGES
8. SCREENSHOT OF REBOOT PAGE
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GXP-2000 Users Manual
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Welcome
Thank you for purchasing the Grandstream GXP – 2000 SIP Enterprise Phone, an affordable, high-quality
IP phone designed for the small to large business enterprise.
Grandstream GXP-2000 is a next-generation enterprise SIP telephone that is feature rich, easy to use,
supports Power-over-Ethernet. The GXP-2000 is ideal for the small and medium size business seriously
looking to leverage their broadband network. Leading industry experts consistently recognize the GXP-
2000 as best in class. It recently received the prestigious 2006 “Excellence Award” from Internet
Telephony Magazine.
The GXP-2000 features intuitive user interfaces, four(4) individual lines, dual 10/100mbps Ethernet ports,
graphical LCD display and a secure central configuration. This SIP phone combines feature functionality
with the latest technology to offer excellent audio quality, ease of use, expandability, and broad
interoperability with 3rd party SIP platforms. It is ideal for any business communication environment.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation
of this product in any way other than as detailed by this User Manual, could void your manufacturer
warranty.
•
•
•
This document is contains links to Grandstream GUI Interfaces. Please download these examples
This document is subject to change without notice. The latest electronic version of this user manual
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,
for any purpose without the express written permission of Grandstream Networks, Inc. is not
permitted.
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GXP-2000 Users Manual
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Installation
EQUIPMENT PACKAGING
The GXP-2000 phone package contains:
1) One GXP-2000 Main Case
2) One Handset
3) One Phone Cord
4) One Universal Power Adaptor
5) One Ethernet Cable
The GXP-Extension package contains:
6) One GXP-Extension unit
7) One PS2 cable
8) One connection plate
9) One Universal Power Adaptor
CONNECTING YOUR PHONE
Figure 1: Connecting the GXP-2000
GXP Ext
Connection
WAN/LAN RJ-45
10/100Mbps ports
Power
Jack
Headset Jet
Table 1: Definitions of the GXP- 2000 Connectors
GXP
Ext Connection
Connect the GXP Ext directly to the GXP using connection cable. Draws power
from PoE.
10/100Mbps RJ-45 ports for WAN (PC) and LAN connections (switched or routed).
Support PoE (802.3af). Draws power from both spare line and signal line
WAN/LAN
5V DC power port; UL Certified
2.5mm Headset port
Power Jack
Headset Jack
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WALL MOUNT
GXP-2000 can be wall mounted. To position the phone on the wall, place two fixed hangers on the wall,
hang the back of the phone on the fixed hangers.
Top Wall
Mount hole
Bottom Wall
Mount hole
To use the handset, pull out the tab (extension downward) from the handset cradle, rotate the tab and plug it
back into the slot with the extension up to hold the handset.
Tab with
extension
Handset
Rest
Tab with
extension up
Tab
GXP EXTENSION SIDE CAR
GXP-2000 supports two (2) extension side cars, providing up to 112 additional programmable extensions.
Each GXP Extension has 56 multi-purpose keys, dual color LEDs (red/green) and support BLF (busy
lamp field) and BLA (bridged line appearance). Connect the first GXP -EXT to the GXP-2000 using the
PS2 cable found in the GXP Ext package. The first GXP-Ext draws power directly from the phone.
Connect the second GXP Ext using the connection plate and the PS2 cable. The GXP2000 will
automatically reboot and power up the GXP Extensions. Grandstream recommends, though not required,
to use a separate power supply with the second GXP Ext. NOTE: should your system loose power,
please unplug your devices and power up the GXP-2000 first.
Powering up the system:
1. The GXP-2000 will boot up first;
2. The GXP LEDs will be solid red;
3. The status light in the top right corner of the GXP-Ext will blink red;
4. All of the LED indicators on the GXP-Ext will flash three times;
5. The status light at the top right corner of the GXP-Ext will turn to solid green.
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Figure 2: Connecting the GXP-2000 and the GXP-Extension
GXP-2000 w/GXP-Extension
GXP Extension
Connecting the GXP-2000
w/GXP-Extension
Reverse side of connection
w/connection plate
FIGURE 3: GXP-2000 INTERNAL HEADSET WIRING SCHEMA
NOTE: A 3.5mm to 2.5mm plug converter is required to use a 2.5mm headset. Purchase a converter at
any electronics store.
SAFETY COMPLIANCES
The GXP-2000 phone complies with FCC/CE and various safety standards. The GXP-2000 power
adaptor is compliant with UL standard. Only use the universal power adapter provided with the GXP
package. The manufacturer’s warranty does not cover damages to the phone caused by unsupported
power adaptors.
WARRANTY
If you purchased your GXP from a reseller, please contact the company where you purchased your
phone for replacement, repair or refund. If you purchased the product directly from Grandstream, contact
your Grandstream Sales and Service Representative for a RMA (Return Materials Authorization) number
before you return the product. Grandstream reserves the right to remedy warranty policy without prior
notification.
Warning: Please do not use a different power adaptor with the GXP-2000 or the GXP-Extension as
they may cause damage to the products and void the manufacturer warranty.
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GXP-2000 Users Manual
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Product Overview
Back View
Desk View
Side View
Table 2: GXP Key Features in a Glance
Feature
Benefit
Open Standards Compatible
SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP,
DNS (A record and SRV), DHCP (both client and server), PPPoE,
TFTP, NTP, Telnet, and TLS.
Network Interfaces
Superb Audio Quality
Feature Rich
Dual 10/100mbps Ethernet ports; 2 USB (2.0) host ports, headset
jack.
Advanced Digital Signal Processing (DSP), Silence suppression,
VAD, CNG, AGC.
Standard voice features including caller ID, call waiting, hold, transfer,
forward, block, mute, autodial, off-hook dial, and click to dial.
Multi-line support, multi-party conferencing, up to 112 additional
extensions, headset enabled, intercom, AES encryption.
Custom down-loadable ring-tones, Multi-language support (MLS),
XML enabled, SRTP & TLS (pending)
Advanced Functionality
Advanced Features
Table 3: Hardware Specifications
Feature / Model
GXP- 2000
LAN Interface (Ethernet ports)
Two (2) 10/100 Full/Half Duplex Ethernet Switch with LAN and PC port
with auto detection
Graphic LCD Display
Expansion Module Support
Headset Jack
8 line x 22 character; 64 rows x 131 columns in pixels
Yes
2.5mm Headset port
LED
11 LED with different light pattern in dual color (g/r)
Power over Ethernet
Built-in, autosensing: Cisco and IEEE 802.3af standard: can
draw power from both spare lines or signal lines from Ethernet
Universal Switching
Power Adaptor
Dimension
Weight
Temperature
Humidity
Input: 100-240VAC 50-60 Hz
Output: +5VDC, 1200mA, UL certified
220mm(l) x 215mm(w) x 57mm(h)
0.82kg (1.8lbs)
32–104oF/ 0–40oC
10% - 90% (non-condensing)
FCC / CE / C-Tick
Compliance
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Table 4: GXP-2000 Technical Specifications
Lines
4 direct lines with 7 speed dial keys; up to 11 line calls (with an additional 112 lines
with 2 daisy-chained GXP-2000 Ext)
Protocol
Support
Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP,
ARP/RARP, ICMP, DNS, DHCP, NTP/SNTP, TFTP, SIMPLE/PRESENCE protocols
Support multiple SIP accounts and up to 11 media channels concurrently
Support SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856,
3863) for use of 7 MFKs and GXP-2000EXT, SIP Dialog package (RFC 4235)
Support for SIP MESSAGE method (RFC 3428)S
Stores up to 100 incoming IM messages (drops IM message 101 plus)
8 line x 22character, 64 rows x 130 column in pixels
Display
Feature Keys
8 dedicated keys: Message Button, Hold, Transfer, Conference, Speakerphone,
Send, Mute/Del, 5 display/menu navigation keys, dual color LEDs
Device
Management
NAT-friendly remote software upgrade (via TFTP/HTTP) for deployed devices
including behind firewall/NAT, Auto/manual provisioning system, GUI Interface,
Address Book
Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
Support for GXP-2000 Extension console and diagnostic mode for keys on GXP-
2000 Extension console
Audio Features
Full-duplex hands-free speakerphone, headset enabled
Advanced Digital Signal Processing (DSP)
Dynamic negotiation of codec and voice payload length
Support for G.723,1 (6.3K), G.729A/B, G.711 µ/A, G.726 (40K/32K/24K/16K),
G.728, G.722 (wide-band), GSM and iLBC codecs
In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Silence Suppression, VAD (voice activity detection), CNG (comfort noise
generation), ANG (automatic gain control)
Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for
speakerphone mode
Support side tone, Adaptive jitter buffer control (patent-pending) and packet delay &
loss concealment
Telephony
Features
Intuitive graphic user interface (GUI), downloadable phone book (XML, LDAP), MLS
(multi language support)
Voice mail indicator with indicator, downloadable custom ring-tones
Call hold, call transfer (attended/blind)
Do-Not-Disturb (DND), call forwarding, call waiting, call waiting caller ID, mute,
redial, call log, and volume control, caller ID display or block
Transfer, hold, forward, multi-party conferencing, dial plans, off-hook auto dial, auto
answer, early dial and speed dial. Support for anonymous call using privacy header
Network and
Provisioning
Via kepad/LCD, Web browser, or secure (AES encrypted) central configuration file,
manual or dynamic host configuration protocol (DHCP) network setup
Support NAT traversal using IETF STUN and Symmetric RTP
Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS
Firmware
Upgrades
Support firmware upgrade via TFTP or HTTP, Support for Authenticating
configuration file before accepting changes
User specific URL for configuration file and firmware files
Advanced
Server Features
Message waiting indication, support DNS SRV Look up and SIP Server Fail Over,
Support customizable idle screen via downloading XML by HTTP/TFTP
Security
DIGEST authentication and encryption using MD5 and MD5-sess, SRTP over TLS
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GXP-2000 Users Manual
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Using GXP-2000 SIP Enterprise Phone
GETTING FAMILIAR WITH LCD
Do Not
Disturb
(blinks)
Handset
enabled
Speaker -phone
enabled
Volume
Calls
Forwarded
Connectivity Status
and/or
SIP Proxy/Server status
Time
User ID: will show
Date
Name
Extension
Use any
picture or logo
Phone IP Address
(located here after boot-up)
Table 5: LCD Icon Definitions
Icon
LCD Icon Definitions
Connectivity Status / SIP Proxy/Server Icon:
Solid – connected to SIP Server/IP address received
Blinking – physical connection failed
Blank – SIP Proxy/Server not registered
Phone Status Icon:
OFF when the handset is on-hook
ON when the handset is off-hook
Speaker Phone Status Icon:
FLASH when phone rings or a call is pending
OFF when the speakerphone is off
ON when the speakerphone is on
DND Icon:
ON when the “do not disturb” is activated
Activate by pressing MUTE/DEL button twice
Calls Forwarded Icon:
INDICATES calls are forwarded
Follow ‘call forwarding’ procedures
Handset, Speakerphone and Ring Volume Icon:
Each icon appears next to the volume icon
To adjust volume, use the up/down button
Real-time Clock:
Synchronized to Internet time server
Time zone configurable via web browser
AM/PM indicator
AM
PM
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Figure 4: Getting Familiar with Keypad
Adjustable LCD screen
Message Waiting Indicator
Line 1-4 Keys
Menu Keys
Mute/Delete
Message
Conference
Transfer
Speed Dial / Configurable
line indicators
RJ11
Hold
Speaker Send/Re-Dial Standard Keypad
Table 6: GXP Button Definitions
Key Button
LINE1-LINE4
Key Button Definitions
4 Line keys with LED, can be configured to different SIP profiles
Use button to select next “Menu Item” when phone is in keypad configuration
mode; Or increase handset/speakerphone volume when phone is ACTIVE; Or
increase ring volume when phone is in IDLE mode
UP ↑
Use button to select previous “Menu Item” when phone is in keypad configure
mode; Or decrease handset/speakerphone volume when phone is ACTIVE;
Or decrease ring volume when is in IDLE mode
DOWN ↓
LEFT Å
Shift cursor to left
Shift cursor to right
RIGHT Æ
Enter Keypad Configuration “MENU” mode when phone is in IDLE mode. Use
as ENTER key when in Keypad Configuration.
OK
TRNF
TRANSFER key: Transfer an ACTIVE call to another number
Press CONF button to connect Calling/Called party into conference
Enter to retrieve (video) voice mails or other messages
Enable/Disable hands-free speaker mode
CONF
MSG
SPEAKER
HOLD
Place ACTIVE call on hold
Press SEND to dial a new number or redial the last number dialed. Press
send button to send a call immediately before “no key entry timeout” value
expires
SEND
Standard phone keypad; press # key to send call; press * key to for IVR
functions
0 - 9, *, #
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COMPLETING AND ANSWERING PHONE CALLS
Handset, Speakerphone and Headset Mode
Handset can be toggled between Speaker or Headset. To switch between Handset and
Speaker/Headset, press the Hook Flash in the handset cradle or press the SPEAKER button.
Multiple SIP Accounts and Lines
The GXP-2000 can support up to 4 independent SIP accounts. Each account is capable of independent
SIP server, user and NAT settings. Each of the 4 line buttons (LINE1-LINE4) is “virtually” mapped to an
individual SIP account. In off-hook state, select an idle line and the name of the account (as configured in
the web interface) is displayed on the LCD and a dial tone is heard.
For example: Configure the 4 SIP accounts as “FreeWorldDialup”, “BroadVoice” , “FreeWorldDialup” and
“Asterisk PBX” respectively and ensure each is active and registered. When LINE1 is pressed, you will
hear a dial tone and see “FreeWorldDialup” on the LCD display; when LINE2 is pressed, you will hear a
dial tone and see “BroadVoice” on the LCD display; when LINE3 is pressed, you will hear a dial tone and
see “Asterisk PBX” on the LCD display.
To make a call, select the line you wish to use. The LED will light up solid red. Switch lines before dialing
by pressing the same LINE button one or more times. If you continue to press one LINE, the selected
account will circulate among the registered accounts.
For example: when LINE1 is pressed, the LCD displays “FreeWorldDialup”; If LINE1 is pressed twice, the
LCD displays “BroadVoice” and the subsequent call will be made through SIP account 2 - BroadVoice.
Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When
the “virtually” mapped line is in use, the GXP-2000 will flash the next available LINE (from left to right) in
red.
A line is ACTIVE when it is in use and the corresponding LED is solid red.
Completing Calls
There are four ways to complete a call:
1. DIAL: Take Handset/SPEAKER/Headset off-hook or press an available LINE key (activates
speakerphone). The line will have a dial tone and the primary line (LINE1) LED is red. If you
wish, select another LINE key (alternative SIP account). Enter the phone number and press the
SEND key.
2. REDIAL: Take Handset/SPEAKER/Headset off-hook or press an available LINE key (activates
speakerphone), the corresponding LED will be red. Press the SEND button to redial the last
number called.
3. USE THE MENU: Press the OK button to bring up the Main Menu. Select Call History and then
Received Calls/Missed Calls/Dialed Calls and select phone number. Press OK to select and
press OK again to dial. The call will dial out in speaker-mode. (Currently applies only to primary
SIP account (LINE 1)).
4. PAGING/INTERCOM: This is only valid if the SERVER/PBX supports this feature and both the
phone and PBX (e.g. Asterisk) are correctly configured. Take the Handset/SPEAKER/Headset
off-hook, select the LINE key associated with account, press OK key to display LCD: LINEx:
PAGE USING. Dial the phone number you want to Page/Intercom and press SEND key.
NOTE: Dial-tone and dialed number display occurs after the handset is off-hook and the line key is
selected. The phone waits 4 seconds (by default; no key entry timeout) before sending and initiating the
call. Press the “SEND” button to override the 4 second delay.
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Making Calls using IP Addresses
Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy. VoIP
calls can be made between two phones if:
•
•
•
both phones have public IP addresses, or
both phones are on a same LAN/VPN using private or public IP addresses, or
both phones can be connected through a router using public or private IP addresses (with
necessary port forwarding or DMZ)
To make a direct IP call, press OK button to bring up MAIN MENU. Select “Direct IP Call”. Press OK to
select. Input the 12-digit target IP address. Press OK key to initiate call.
For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input
the following: 192*168*1*60#5062 - “ * ” key represent “.” and “#” key represent “:”. Press OK to dial out.
Quick IP Call Mode
Dial an IP address under the same LAN/VPN by keying in the last octet in the IP address. This simulates
a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended.
In the “Advanced Settings” page, set the "Use Quick IP-call mode to YES. When #xxx is dialed, where x
is 0-9 and xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. “aaa.bbb.ccc” is from the local IP
address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required
(but OK).
For example:
192.168.0.2 calling 192.168.0.3 -- dial #3 follow by SEND or #
192.168.0.2 calling 192.168.0.23 -- dial #23 follow by SEND or #
192.168.0.2 calling 192.168.0.123 -- dial #123 follow by SEND or #
192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3
NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct
IP-IP call will also use STUN. Configure the “Use Random Port” to “NO” when completing Direct IP calls.
Receiving Calls
1. Incoming single call: Phone rings with selected ring-tone. The corresponding account LINE
flashes red. Answer call by taking Handset/SPEAKER/Headset off hook or pressing SPEAKER.
2. Incoming multiple calls: Call Waiting tone (stutter tone) audible. Next available lines will flash
red (as described in section 4.3.2).
3. Paging/Intercom Enabled: Phone beeps once and automatically establishes the call via
SPEAKER. (PBX (or Server) must also supports this feature)
Call Waiting/ Call Hold
1. Hold: Place a call on ‘hold’ by pressing the “HOLD” button.
2. Resume: Resume call by pressing the corresponding blinking LINE.
3. Multiple Calls: Automatically place ACTIVE call on ‘HOLD’ by selecting another available LINE
to place or receive another call. Call Waiting tone (stutter tone) audible when line is in use.
Call Transfer
GXP-2000 supports both blind and attended (or supervised) transfer:
1. Blind Transfer: Press “TRNF” button, then dial the number and press the “SEND” button to
complete transfer of active call.
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2. Attended (or Supervised) Transfer: Press “Linex” button to make a call and automatically
place the ACTIVE LINE on HOLD. Once the call is established, press “TRNF” key to transfer the
call and hang up.
NOTE: To transfer calls across SIP domains, SIP service providers must support transfer across SIP
domains. Blind transfer will usually use the primary account SIP profile.
3-Way Conferencing
GXP-2000 supports 3-way conferencing.
1. Initiate a Conference Call: Place first call(er) on HOLD, call second party. Press the CONF
button then the LINE that is on HOLD (blinking). Repeat for third party.
2. Cancel Conference: If after pressing the “CONF” button, a user decides not to conference
anyone, press CONF again or the original LINE button. This will resume two-way conversation.
3. End Conference: To end a conference, press HOLD. This breaks the conference and places
both parties on hold. To speak with an individual party, select the corresponding blinking LINE.
Check Messages (Message Waiting Indicator)
The blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Press the MSG button
to retrieve the message. An IVR will prompt the user through the process of message retrieval. Press a
specific LINE to retrieve messages for a specific line account.
NOTE:
•
Each line has a separate voicemail account. Each account requires a voicemail portal number to
be configured in the “voicemail user id” field.
•
To check which line account has a message – 1) press the message button (this always checks
the primary account), 2) check each line for stutter tone or 3) check missed calls using the menu.
Mute/Delete
1. To enable/disable mute during a call, press “MUTE/DEL” button. The red Mute/Delete icon
(muted microphone) will flash on the LCD. Press the “MUTE/DEL” button to resume audio.
2. To delete entries while using the “MAIN MENU” place cursor (use the arrows to move cursor)
before the entry. Press “MUTE/DEL” to delete the digit.
Do Not Disturb
1. Press the “Mute/Delete” button if you do not want to take a call. This will send the caller directly
to voicemail.
2. Press the “Mute/Delete” button to set phone to ‘do not disturb’ (icon will be on the screen). The
phone will not ring and send caller directly to voicemail. (see note above)
Speed Dial
The seven multi-functional buttons can be configured for speed dial. Press the speed dial button to
automatically call the assigned extension.
Note: The multi-functional buttons will function as LINE keys when all four primary lines are busy. The
LED will flash indicating the incoming call. Press the button to pick up the call. If any one of the 7
functions keys is associated with a call, the speed dial/BLF function will not work. For example: when
first multi-functional button is in use, you cannot use it for speed dial/BLF.
Asterisk Busy Line Field
The seven multi-functional buttons can be configured for Asterisk Busy Line Field function with a specified
account. When Asterisk BLF is configured on one of the multi-functional buttons, the Speed Dial function
will work when it that line is not in use.
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Extension Board
The fifty-six multi-functional buttons on GXP-Extension board function the same as multi-functional
buttons on the phone base station. These extensions are fully programmable but can not be used as line
keys.
CALL FEATURES
The GXP-2000 supports traditional and advanced telephony features.
Table 7: GXP-2000 Call Features
Key
*30
*31
*67
*82
*70
*71
*72
Call Features
Block Caller ID (for all subsequent calls)
Send Caller ID (for all subsequent calls)
Block Caller ID (per call)
Send Caller ID (per call)
Disable Call Waiting (per Call)
Enable Call Waiting (per Call)
Unconditional Call Forward
Dial “*72” for a dial tone. Dial the forwarding number followed by “#”. Wait
for dial tone. LCD will display “Call FWD Activated”.
Cancel Unconditional Call Forward: dial “*73” and get the dial tone, then
hang up. LCD will display “Call FWD Activated”.
Busy Call Forward
*73
*90
Dial “*90” for a dial tone. Dial the forwarding number followed by “#”. Wait
for a dial tone. Hang up.
*91
*92
Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up.
Delayed Call Forward
Dial “*92” for a dial tone. Dial the forwarding number followed by “#”. Wait
for a dial tone. Hang up. LCD will display “Call FWD Activated”.
Cancel Delayed Call Forward
*93
Dial “*93” for a dial tone, then hang up.
CUSTOMIZED LCD SCREEN & XML
The GXP-2000 supports both a customized LCD idle-screen and a downloadable XML phonebook. Detailed
information and configuration guides are located on the website @ http://www.grandstream.com/y-
downloads.htm.
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CONFIGURATION GUIDE
CONFIGURATION WITH KEYPAD
To enter the MENU, press the round button. Navigate the menu by using the arrow keys: up/down and
left/right. Press the OK button to confirm a menu selection, delete an entry by pressing the MUTE/DEL button.
The phone automatically exits MENU mode with an incoming call, the phone is off-hook or the MENU mode if
left idle for 20 seconds.
Table 8: Key Pad Configuration Menu
Display “Call History”
Press Menu button to enter this menu including
“Received Calls” or
“Dialed Calls” or
“Missed Calls” or
“Back”
Press ‘↓’ or ’↑’ to toggle the selection
Press ‘←’ to return to the main menu
Display “Status”
Press Menu button to enter this menu to see the status of the phone
Press ‘↓’ or ’↑’ to toggle the selection
Press Menu or ‘←’button to exit
Display “Phone Book”
Press Menu button to display the phone book including
“Download Phonebook”
Press ‘↓’ or ’↑’ to toggle the selection
Press Menu button to choose the menu item
Press ‘←’ button to return to the main menu
Display “Instant Messages”
Press Menu button to display the Instant Messages received.
Press ‘↓’ or ’↑’ to toggle the selection
Press Menu button to choose the menu item
Press ‘←’ button to return to the main menu
Note: GXP-2000 only supports the function of receiving of Instant
Messages.
Display “Direct IP Call”
Display “Preference”
Press Menu button to display the direct IP call interface
Enter the 12 digit IP address. For example, enter the IP address:
10.10.1.2 like 010010001002.
Press ‘←’ or ‘→’ to move the cursor or toggle the selection
Press Menu button to confirm.
Press Menu button to enter this sub menu including
“Do NOT Disturb” or
“Ring Tone” or
“Ring Volume” or
“Download SCR XML” or
“Erase Custom SCR” or
“Back”
DND (Do NOT Disturb) function could be turned on or off in the “DO NOT
Disturb” menu.
Choose different ring tones in the “Ring Tone” menu.
Press ‘←’ or ‘→’ to adjust ring volume in the “Ring Volume” menu.
Press ‘↓’ or ’↑’ to toggle the selection.
Press Menu button to choose the menu item.
Press ‘←’ to return to the main menu.
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Display “Configure”
Press Menu button to display the configuration selections:
“Network” or
“SIP” or
“Audio” or
“Upgrade” or
“Factory Reset”
More detailed information on these selections are in the web
configuration page.
Press ‘↓’ or ’↑’ to toggle the selection.
Press ‘←’ to return the main menu.
Display “Factory Functions”
Press Menu to display the factory function items including
“Ethernet Loopback” or
“Audio Loopback” or
“Diagnostic Mode” or
“Enable Diag Port” or
“Back”
Press ‘↓’ or ’↑’ to toggle between selections.
Press ‘←’ to return to the main menu.
Display “Reboot”
Press Menu button to reboot the device
Display “Exit”
Press Menu button to exit the menu
Display “Ring Volume”
Press Menu button to hear the selected ring volume, press ‘←’ or ’ →’ to
hear and adjust the ring tone volume.
Press Menu button to select and exit, take effect immediately.
Display “Ethernet Loopback” Press Menu button to enter this mode
Connect a cross Ethernet cable from your “PC” port, and the “LAN” port.
The test result is displayed on the screen. Use this feature to diagnose
the state of health of the RJ45 jacks.
Press Menu button to exit the diagnostic mode.
Note: Running the Ethernet Loopback mode with a normal connection
will cause IP loss.
Display “Audio Loopback”
Display “Diagnostic Mode”
Display “Factory Reset”.
Press Menu button to enter this mode
Speak into the handset. If you hear your voice in the handset, your audio
works fine.
Press Menu button to exit the mode.
Press Menu button to enter this mode, all LEDs will light up
Press any key on the keypad, to display the button name in the LCD. Lift
and put back the handset or press Menu button to exit the diagnostic
mode.
Key in the physical/MAC address on back of the phone.
Press Menu button to reset FACTORY DEFAULT setting. Do not use
Factory Reset unless you want to restore factory settings
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CONFIGURATION WITH WEB BROWSER
The GXP-2000 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded
HTML pages allow a user to configure the IP phone through a Web browser such as Microsoft’s IE or
Mozilla Firefox.
ACCESS THE WEB CONFIGURATION MENU
is the IP address of the phone, which is displayed on the LCD screen.
END USER CONFIGURATION
After requesting HTTP, you will see a login screen. The password is case sensitive with maximum length
of 25 characters. The factory default password for the End User is “123”, for Administrator is “admin”.
Only the administrator has privileges to access the Advanced Setting and Account information. A non-
administrator will receive an error message.
Table 9: Device Configuration – Basic Settings Definitions
End User
Password
This contains the password to access the Web Configuration Menu. This field is
case sensitive with a maximum length of 25 characters.
IP Address
There GXP-2000 operates in two modes:
1. DHCP mode: all the field values for the Static IP mode are not used (even
though they are still saved in the Flash memory.) The GXP-2000 acquires
its IP address from the first DHCP server it discovers on its LAN. The
DHCP option is reserved for NAT router mode. To use the PPPoE feature,
set the PPPoE account settings. The GXP-2000 establishes a PPPoE
session if any of the PPPoE fields are set.
2. Static IP mode: configure all of the following fields: IP address, Subnet
Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2
(secondary). These fields are set to zero by default.
Time Zone
This parameter controls the date/time display according to the specified time zone.
Daylight Savings Time
This parameter controls time displayed in daylight savings time or not. If set to
“Yes”, then the displayed time will be 1 hour ahead of normal time.
Time Display Format
Date Display Format
LCD time display in 12 hour or 24 hour format
Choose one of the following formats:
•
•
•
Year-Month-Day
Month-Day-Year
Day-Month-Year
Display Clock instead of Default is No. Display the clock instead of Date. If set to “Yes”, it will show the
Date
clock.
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Table 10: Device Configuration - Status Definitions
Hardware Revision
MAC Address
IP Address
Hardware version number: Main Board, Interface Board
The device ID, in HEX format. This is a very important ID for ISP troubleshooting.
This field shows IP address of GXP-2000
Product Model
Software Version
This field contains the product model information.
•
Program: This is the main software (firmware) release number, always used to
identify the software (firmware) system of the phone.
Loader: Driver loader code version number.
•
•
Boot: Booting code version number
System Up Time
Registered
This field shows system up time since the last reboot.
Iindicates whether accounts are registered to the related SIP server(s). GXP-2000
can support four unique SIP profiles.
PPPoE Link Up
Inidicates whether the PPPoE connection is enabled (connected to a modem).
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ADVANCED USER CONFIGURATION
To login to the Advanced User Configuration page, please follow the instructions in section 5.2.1. The
password is case sensitive with a maximum length of 25 characters and the factory default password for
Advanced User is “admin”.
Advanced User configuration includes not only the end user configuration, but also advanced
configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous
configuration.
Table 11: Advanced User Configuration Page Definitions
Admin
Password
Administrator password. Only the administrator can configure the “Advanced Settings”
page. Password field is purposely blank for security reasons after clicking update and
saved. The maximum password length is 25 characters.
Silence
Suppression
This controls the silence suppression/VAD feature of the audio codec G.723 and G.729.
If set to “Yes”, when silence is detected, a small quantity of VAD packets (instead of
audio packets) will be sent during the period of no talking. If set to “No”, this feature is
disabled.
Voice Frames
per TX
This field contains the number of voice frames to be transmitted in a single Ethernet
packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500
byte (or 120kbps)).
When setting this value, be aware of the requested packet time (ptime, used in SDP
message) is a result of configuring this parameter. This parameter is associated with the
first codec in the above codec Preference List or the actual used payload type
negotiated between the 2 conversation parties at run time. e.g., if the first codec is
configured as G.723 and the “Voice Frames per TX” is set to 2, then the “ptime” value in
the SDP message of an INVITE request will be 60ms because each G.723 voice frame
contains 30ms of audio. Similarly, if this field is set to 2 and the first codec is G.729 or
G.711 or G.726, then the “ptime” value in the SDP message of an INVITE request will be
20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the IP phone
will use and save the maximum allowed value for the corresponding first codec choice.
The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20 (x10ms) frames;
for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames
respectively.
Please be careful when editing these parameters. Adjusting these parameters will also
change the dynamic jitter buffer. The GXP-2000 has a patent dynamic jitter buffer
handling algorithm. The jitter buffer range is 20 ~ 200 ms.
Grandstream recommends using the default settings provided. Grandstream does not
recommend adjusting these parameters if you are an average user. Incorrect settings
will affect the voice quality. Please refer to the Codec FAQ at
http://www.grandstream.com/FAQ/FAQ-Codec.pdf for more technical detail.
Layer 3 QoS
Layer 2 QoS
This field defines the layer 3 QoS parameter. It is the value used for IP Precedence or
Diff-Serv or MPLS. Default value is 48.
This contains the value used for layer 2 VLAN tag. Default setting is blank.
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No Key Entry
Timeout
Default is 4 seconds.
Use # as
Dial Key
This parameter allows users to configure the “#” key as the “Send” (or “Dial”) key. If set
to “Yes”, the “#” key will immediately send the call. In this case, this key is essentially
equivalent to the “(Re)Dial” key. If set to “No”, the “#” key is included as part of the dial
string.
Local RTP port This parameter defines the local RTP-RTCP port pair used to listen and transmit. It is the
base RTP port for channel 0. When configured, channel 0 will use this port _value for
RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and
port_value+3 for its RTCP. The default value is 5004.
Use Random
Port
This parameter, when set to “Yes”, will force random generation of both the local SIP
and RTP ports. This is usually necessary when multiple GXP-2000s are behind the
same NAT. Default is No.
Keep-alive
interval
This parameter specifies how often the GXP-2000 sends a blank UDP packet to the SIP
server in order to keep the “hole” on the NAT open. Default is 20 seconds.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
STUN Server
IP address or Domain name of the STUN server. STUN resolution result will display in
the STATUS page of the Web UI.
Firmware
Upgrade
This radio button will enable GXP-2000 to download firmware or configuration file
through either HTTP or local TFTP server. Choices are mutually exclusive.
Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero or not
blank, the GXP-2000 will attempt to retrieve a new configuration file or new code image
from the specified TFTP server at boot time. It will make up to 3 attempts before timeout
and then it will start the boot process using the existing code image in the Flash memory.
If a TFTP server is configured and a new code image is retrieved, the new downloaded
image will be verified and then saved into the Flash memory.
Note: Grandstream strongly recommends that the user upgrade firmware locally in a
LAN environment if using TFTP to upgrade. Please do NOT interrupt the TFTP upgrade
process (especially the power supply) as this will damage the device.
Via HTTP
Server
The HTTP server URL used for firmware upgrade and configuration via HTTP. For
Here “:6688” is the specific TCP port that the HTTP server is using; omit if using default
port 80.
Note: If Auto Upgrade is set to No, GXP-2000 will only perform HTTP download once at
boot up.
Automatic
Upgrade
This function is used by ITSP. End user should NOT touch these parameters.
Default is No. Choose “Yes” to enable automatic HTTP upgrade and provisioning.
In “Check for upgrade every” field, enter the number of minutes to check the HTTP
server for firmware upgrade or configuration changes. When set to “No”, the phone will
only perform HTTP upgrade and configuration check once at boot up.
Phonebook
XML
Enable the XML phonebook via TFTP or HTTP. Define XML server path and download
speed.
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Idle Screen XML Enable XML Idle Screen via TFTP or HTTP. Define XML server path.
DTMF Payload
Type
This parameter sets the payload type for DTMF using RFC2833. Default is 101.
Syslog Server
Syslog Level
The IP address or URL of System log server. This feature is especially useful for ITSPs.
Select the ATA to report the log level. Default is NONE. The level is one of DEBUG,
INFO, WARNING or ERROR. Syslog messages are sent based on the following events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level)
NAT related info (INFO level)
sent or received SIP message (DEBUG level)
SIP message summary (INFO level)
inbound and outbound calls (INFO level)
registration status change (INFO level)
negotiated codec (INFO level)
Ethernet link up (INFO level)
SLIC chip exception (WARNING and ERROR levels)
memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains the
following components: GS_LOG: [device MAC address][error code] error message
For example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000].
Ethernet link is up.
NTP server
This parameter defines the URI or IP address of the NTP (Network Time Protocol) serve.
It is used to display the current date/time.
Distinctive Ring Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a particular
Tone
Caller ID. The GXP will ONLY use selected ring tones for particular Caller IDs. For all
other calls, the GXP will use System Ring Tone. When selected and no Caller ID is
configured, the selected ring tone will be used for all incoming calls.
Disable Call
Waiting
Default is No. If set to Yes, the call waiting will be disabled.
Use Quick IP
Call Mode
Dial an IP address under the same LAN/VPN segment by entering the last octet in the IP
address.
In the Advanced Settings page there is an option "Use Quick IP-call mode". Default
setting is No. When set to YES, and #XXX is dialed, where X is 0-9 and XXX <=255,
phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the
local IP address REGARDLESS of subnet mask.
#XX or #X are also valid so leading 0 is not required (but OK). See Quick IP Call Mode
for details.
Lock keypad
update
If set to “Yes”, the configuration changes via keypad are disabled.
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INDIVIDUAL ACCOUNT SETTINGS
The GXP-2000 has 4 line appearances, each with an independent SIP account. Each SIP account
requires its own configuration page. Their configurations are identical.
Table 12: SIP Account Configuration Page Definitions
Account Active
This field indicates whether the account is active. The default value for the
primary account (Account 1) is Yes. The default value for the other two accounts
is No.
Account Name
SIP Server
The name associated with each account - displayed on LCD.
SIP Server’s IP address or Domain name provided by VoIP service provider.
Outbound Proxy
IP address or Domain name of Outbound Proxy, Media Gateway, or Session
Border Controller. Used for firewall or NAT penetration in different network
environment. If the system detects symmetric NAT, STUN will not work. ONLY
outbound proxy can provide solution for symmetric NAT.
SIP User ID
User account information provided by VoIP service provider (ITSP); either an
actual phone number or formatted like one.
Authenticate ID
SIP service subscriber’s Authenticate ID used for authentication. It can be
identical to or different from SIP User ID.
Authenticate Password SIP service subscriber’s account password for GXP-2000 to register to (SIP)
servers of ITSP.
Name
SIP service subscriber’s name that is used for Caller ID display.
Use DNS SRV:
Default is No. If set to “Yes”, the client will use DNS SRV to look up server.
User ID is Phone
Number
If the phone has an assigned PSTN telephone number, this field should be set to
“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be
attached to the “From” header in SIP request
SIP Registration
This parameter controls sending REGISTER messages to the proxy server. The
default setting is “Yes”.
Un-register on Reboot
Register Expiration
Default is No. If set to “Yes”, the SIP user’s registration information will be
cleared on reboot.
This parameter allows user to specify the time frequency (in minutes) that GXP-
2000 refreshes its registration with the specified registrar. The default interval is
60 minutes. The maximum interval is 65,535 minutes (about 45 days).
Local SIP Port
This parameter defines the local SIP port used to listen and transmit. The default
value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and
Account 4 respectively.
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NAT Traversal
This parameter activates the NAT traversal mechanism. If activated (by choosing
“Yes”) and a STUN server is also specified, the phone performs according to the
STUN client specification. Using this mode, the embedded STUN client detects if
and what type of NAT/Firewall configuration is used. If the detected NAT is a Full
Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mapped
public IP address and port in all of its SIP and SDP messages. If the NAT
Traversal field is set to “Yes” with no specified STUN server, the GXP-2000 will
periodically (every 20 seconds or so) send a blank UDP packet (with no payload
data) to the SIP server to keep the “hole” on the NAT open.
Subscribe for MWI:
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Voice Mail User ID
When configured, user can access messages by pressing “MSG” button. This ID
is usually the VM portal access number.
Send DTMF
This parameter specifies the mechanism to transmit DTMF digit. There are 3
supported modes: in audio which means DTMF is combined in audio signal (not
very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
Early Dial
Default is No. Use only if proxy supports 484 response.
Dial Plan Prefix
Enable Call Features
Sets the prefix added to each dialed number.
Default is No. If set to “Yes”, Call transfer, Call Forwarding & Do-Not-Disturb are
supported locally provided ITSP support those features.
Session Expiration
The SIP Session Timer extension enables SIP sessions to be periodically
“refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval
expires, if there is no refresh via a UPDATE or re-INVITE message, the session is
terminated.
Session Expiration is the time (in seconds) at which the session is considered
timed out, provided no successful session refresh transaction occurs beforehand.
The default value is 180 seconds.
Min-SE
Defines the minimum session expiration (in seconds). Default is 90 seconds.
Caller Request Timer
If set to “Yes”, the phone will use session timer when it makes outbound calls if
remote party supports session timer.
Callee Request Timer
Force Timer
If selecting “Yes”, the phone will use session timer when it receives inbound calls
with session timer request.
If set to “Yes”, the phone will use session timer even if the remote party does not
support this feature. If set to “No”, the session timer is enabled only when the
remote party supports this feature. To turn off Session Timer, select “No” for
Caller Request Timer, Callee Request Timer, and Force Timer.
UAC Specify Refresher As a Caller, select UAC to use the phone as the refresher, or UAS to use the
Callee or proxy server as the refresher.
UAS Specify Refresher As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to
use the phone as the refresher.
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Force INVITE
Enable 100rel
Session Timer can be refreshed using INVITE method or UPDATE method.
Select “Yes” to use INVITE method to refresh the session timer.
PRACK (Provisional Acknowledgment) method enables reliability to SIP
provisional responses (1xx series). This is required to support PSTN inter-
networking..
Account Ring Tone
There are 4 uniquely defined ring tones:
•
One (1) System Ring Tone: when selected, all calls will ring with system
ring tone.
•
Three (3) Customer Ring Tones: when selected, incoming calls from
designated account will play selected ring tone.
Send Anonymous
Auto Answer
If this parameter is set to “Yes”, the “From” header in outgoing INVITE message
will be set to anonymous, essentially blocking the Caller ID from displaying.
Default is No. If set to “Yes”, GXP-2000 will automatically switch on speaker to
answer the incoming call. Set to Intercom/Paging mode, it will answer the call
based on the SIP info header from the server.
Preferred Vocoder
The GXP-2000 supports up to 5 different Vocoder types including G.711(a/µ)
(also known as PCMU/PCMA), GSM, G.723.1, G.729A/B.
Configure Vocoders in a preference list that is included with the same preference
order in SDP message. Enter the first Vocoder in this list by choosing the
appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by
choosing the appropriate option in “Choice 8”.
Jitter Delay
Select desired Jitter Buffer Delay. Default is Medium. Grandstream recommends
“High” for a poor network environment.
Special Feature
Default is Standard. Choose the selection to meet special requirements from Soft
Switch vendors.
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Saving the Configuration Changes
After the user makes a change to the configuration, press the “Update” button in the Configuration Menu.
The LCD will then display the following screen to confirm saved changes.
Grandstream recommends to power cycle the IP phone after saving changes.
Rebooting the Phone from Remote
Press the “Reboot” button at the bottom of the configuration menu to reboot the phone remotely. The LCD
will then display the following screen to confirm reboot is underway.
Wait 30 seconds to log-in again.
CONFIGURATION THROUGH SECURE CENTRAL PROVISIONING SERVER
The end-user can automatically configure one or more GXP-2000 from a secure central provisioning
system by downloading the configuration file from the central server. The format of the configuration file
is as follows: “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP-2000.
GAPSLITE - Grandstream Automated Provisioning System – supports the automated configuration of the
GXP-2000. It is licensed-based software. GAPSLITE uses enhanced (NAT friendly) TFTP or HTTP and
other communication protocols to communicate with each individual GXP-2000 for firmware upgrade,
remote reboot, etc. The GAPSLITE software package also has a configuration tool to generate device
configuration files.
GAPSLITE is the default for all Grandstream devices. Based on the unique MAC address, GAPSLITE
provisions the devices with re-direction settings to point to a customer’s TFTP or HTTP server for further
provisioning. This could be simple re-direction or with special provisioning settings.
The GAPSLITE configuration tool is free with purchases over 512 units. Under 512 units, the license fee
is $99.95. The tool and configuration templates is available on http://www.grandstream.com/y-
configurationtool.htm
Please refer to GAPSLITE product documentation or contact Grandstream Sales Department for more
information.
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Software Upgrade & Customization
Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding
configuration settings are in the ADVANCED SETTINGS configuration page.
Firmware Upgrade through TFTP/HTTP
To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. “Upgrade Server” needs to be set
to a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are
examples of some valid URLs.
•
•
firmware.mycompany.com:6688/Grandstream/1.0.0.18
168.75.215.189
There are two ways to set up the Upgrade Server to upgrade firmware: Key Pad Menu and Web
Configuration Interface.
Key Pad Menu
To configure the Upgrade Server via Key Pad Menu options, select “Config” from the Main Menu, then
select “Upgrade”. Under this sub Menu, user can edit Upgrade Server in either an IP address format or
FQDN format. Choose “Save and use TFTP” or “Save and use HTTP” to select upgrade method. Select
“Reboot” from the Main Menu to reboot the phone.
Web Configuration Interface
To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the
GXP-2000 IP address. Enter the admin password to access the web configuration interface. In the
ADVANCED SETTINGS page, enter the Upgrade Server’s IP address or FQDN in the “Upgrade Server”
field. Select TFTP or HTTP upgrade method. Update the change by clicking the “Update” button.
“Reboot” or power cycle the phone to update the new firmware.
During this stage, the LCD will display the firmware file downloading process. If a firmware upgrade fails
for any reason (e.g., TFTP/HTTP server is not responding, there are no code image files available for
upgrade, or checksum test fails, etc), the phone will stop the upgrading process and re-boot using the
existing firmware/software.
Firmware upgrades take around 20 seconds in a controlled LAN or 2-3 minutes over the Internet.
Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever
possible.
No Local TFTP Server
For users who do not have local TFTP server, Grandstream provides a NAT-friendly TFTP server on the
public Internet for users to download the latest firmware upgrade automatically. Please check the
Support/Download section of our website to obtain this TFTP server IP address:
Alternatively, download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades.
customerFree.cfm.
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Directions to configure local TFTP:
1. Unzip the file and put all of them under the root directory of the TFTP server.
2. The PC running the TFTP server and the GXP-2000 should be in the same LAN
segment.
3. Go to File -> Configure -> Security to change the TFTP server's default setting from
"Receive Only" to "Transmit Only" for the firmware upgrade.
4. Start the TFTP server, in the phone’s web configuration page
5. Configure the Firmware Server Path with the IP address of the PC
6. Update the change and reboot the unit
User can also choose to download the free HTTP server from http://httpd.apache.org/ or use Microsoft IIS
web server.
NOTE:
•
When GXP-2000 phone boots up, it will send TFTP or HTTP request to download configuration
file “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP-2000 phone. This
file is for initial provisioning purpose only.
•
For normal TFTP or HTTP firmware upgrades, the following error messages in a TFTP or HTTP
server log can be ignored: “TFTP Error from [IP ADRESS] requesting cfg000b82023dd4 : File
does not exist. Configuration File Download”
NOTE: For earlier firmware versions (prior to 1.1) when the LCD becomes pale or white a power cycle is
required. For hardware version 1.1 or later, the phone automatically reboots after the firmware upgrade is
complete.
CONFIGURATION FILE DOWNLOAD
The GXP-2000 can be configured via Web Interface as well as via Configuration File through TFTP or
HTTP. “Upgrade Server” is the TFTP or HTTP server path for the configuration file. It needs to be set to a
valid URL, either in FQDN or IP address format.
A configuration parameter is associated with each particular field in the web configuration page. A
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page. For a
detailed parameter list, please refer to the corresponding configuration template of the firmware.
Once the GXP-2000 boots up (or re-booted), it will request a configuration file named “cfgxxxxxxxxxxxx”,
where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. The configuration file
name should be in lower cases.
Managing Firmware and Configuration File Download
When “Automatic Upgrade” is set to “Yes”, a Service Provider can use P193 (Auto Check Interval, in
minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at pre-
scheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider
can manage and reduce the Firmware or Provisioning Server load at any given time.
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Restore Factory Default Setting
WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone.
Please backup or print all the settings before you restoring factory default settings. Grandstream is not
responsible for restoring lost parameters and cannot connect to your VoIP service provider.
Directions for Restoration:
Disconnect network cable and power cycle the unit before resetting factory default settings.
1. Step 1: Press “OK” key to bring up the key pad configuration UI menu, select “Config”, press
“OK” to enter submenu, select “Factory Reset” (Please refer to Table 5-1 of keypad flow chart)
2. Step 2: Key in the MAC address printed on the bottom of the sticker. Please use the following
mapping:
0 - 9 : 0 - 9
a. A: 22 (press the “2” key twice, “A” will show on the LCD)
b. B: 222
c. C: 2222
d. D: 33 (press the “3” key twice, “D” will show on the LCD)
e. E: 333
f. F: 3333
For example: if the MAC address is 000b8200e395, it should be key in as “0002228200333395”.
NOTE: If there are digits like “22” in the MAC, you need to type “2” then press “->” right arrow key to
move the cursor or wait for 4 seconds to continue to key in another “2”.
3. Step 3: Press the “OK” key again to move the cursor to “OK” button. Press “OK” key again to
confirm. If the MAC address is correct, the phone will reboot. Otherwise, it will exit to previous
keypad menu interface.
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Glossary of Terms
ADSL Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that transmit
from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800 kbps upstream,
depending on line distance.
AGC Automatic Gain Control is an electronic system found in many types of devices. Its purpose is to
control the gain of a system in order to maintain some measure of performance over a changing range of
real world conditions.
ARP Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically
IPv4, to map IP network addresses to the hardware addresses used by a data link protocol. The protocol
operates below the network layer as a part of the interface between the OSI network and OSI link layer. It
is used when IPv4 is used over Ethernet
ATA Analog Telephone Adapter. Used to convert analog telephone signal in order to use a VoIP data
network.
CODEC Abbreviation for Coder-Decoder. It's an analog-to-digital (A/D) and digital-to-analog (D/A)
converter for translating the signals from the outside world to digital, and back again.
CNG Comfort Noise Generator, generate artificial background noise used in radio and wireless
communications to fill the silent time in a transmission resulting from voice activity detection.
DATAGRAM A data packet carrying its own address information so it can be independently routed from
its source to the destination computer
DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded or
compressed. Lossy compression algorithms ordinarily decimate while sub-sampling.
DECT Digital Enhanced Cordless Telecommunications: A standard developed by the European
Telecommunication Standard Institute from 1988, governing pan-European digital mobile telephony.
DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to the public
switched telephone network, Closed User Groups (CUGs), Local Area Networks, and wireless local loop.
The DECT Common Interface radio standard is a multi-carrier time division multiple access, time division
duplex (MC-TDMA-TDD) radio transmission technique using ten radio frequency channels from 1880 to
1930 MHz, each divided into 24 time slots of 10ms, and twelve full-duplex accesses per carrier, for a total
of 120 possible combinations. A DECT base station (an RFP, Radio Fixed Part) can transmit all 12
possible accesses (time slots) simultaneously by using different frequencies or using only one frequency.
All signaling information is transmitted from the RFP within a multi-frame (16 frames). Voice signals are
digitally encoded into a 32 Kbit/s signal using Adaptive Differential Pulse Code Modulation.
DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain
names into IP addresses
DID Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without going
through an attendant or auto-attendant.
DSP Digital Signal Processor. A specialized CPU used for digital signal processing. Grandstream
products all have DSP chips built inside.
DTMF Dual Tone Multi Frequency. The standard tone-pairs used on telephone terminals for dialing
using in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals
support only 12 of them (0-9, * and #).
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FQDN Fully Qualified Domain Name. A FQDN consists of a host and domain name, including top-level
Grandstream is the second-level domain, and and.com is the top level domain.
FXO Foreign eXchange Office. An FXO device can be an analog phone, answering machine, fax, or
anything that handles a call from the telephone company like AT&T. They should also operate the same
way when connected to an FXS interface.
An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions have their own
standards. FXO is complimentary to FXS (and the PSTN).
FXS Foreign eXchange Station. An FXS device has hardware to generate the ring signal to the FXO
extension (usually an analog phone).
An FXS device will allow any FXO device to operate as if it were connected to the phone company. This
makes your PBX the POTS+PSTN for the phone.
The FXS Interface connects to FXO devices (by an FXO interface, of course).
DHCP The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the
configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP addresses, to
deliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to provide
other configuration information such as the addresses for printer, time and news servers.
echo from a voice communication in order to improve voice quality on a telephone call. In addition to
preventing echo from traveling across a network. There are two types of echo of relevance in telephony:
acoustic echo and hybrid echo. Speech compression techniques and digital processing delay often
contribute to echo generation in telephone networks.
H.323 A suite of standards for multimedia conferences on traditional packet-switched networks.
HTTP Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and retrieve
functions of a server
IP Internet Protocol. A packet-based protocol for delivering data across networks.
IP-PBX IP-based Private Branch Exchange
IP Telephony (Internet Protocol telephony, also known as Voice over IP Telephony) A general term for
the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and
other forms of information that have traditionally been carried over the dedicated circuit-switched
connections of the public switched telephone network (PSTN). The basic steps involved in originating an
IP Telephony call are conversion of the analog voice signal to digital format and compression/translation
of the signal into Internet protocol (IP) packets for transmission over the Internet or other packet-switched
networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony
are often used to mean the same; however, they are not 100 per cent interchangeable, since Internet is
only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP
Telephony software essentially provides free telephone calls anywhere in the world. However, the
challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border
controllers resolve this issue by providing quality assurance comparable to legacy telephone systems.
IVR IVR is a software application that accepts a combination of voice telephone input and touch-tone
keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and
perhaps other media.
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MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-
bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for
Ethernet is 1500 byte.
NAT Network Address Translation
NTP Network Time Protocol, a protocol to exchange and synchronize time over networks The port used
is UDP 123 Grandstream products using NTP to get time from Internet
OBP/SBC Outbound Proxy or another name Session Border Controller. A device used in VoIP networks.
OBP/SBCs are put into the signaling and media path between calling and called party. The OBP/SBC
acts as if it was the called VoIP phone and places a second call to the called party. The effect of this
behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the
OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private
OBP/SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network.
Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks
with internet connections using NAT.
PPPoE Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in
Ethernet frames. It is used mainly with cable modem and DSL services.
PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or
called POT (Plain Old Telephone), or circuit switched network.
Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data, but does not
transport any data itself. It is used periodically to transmit control packets to participants in a streaming
multimedia session. The primary function of RTCP is to provide feedback on the quality of service being
provided by RTP.
RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video
over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first
published in 1996 as RFC 1889
SDP Session Description Protocol is a format for describing streaming media initialization parameters. It
has been published by the IETF as RFC 2327.
SIP Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261).
SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice
transmission and uses fewer resources and is considerably less complex than H.323. All Grandstream
products are SIP based
STUN Simple Traversal of UDP over NATs is a network protocol allowing clients behind NAT (or multiple
NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by
the NAT with a particular local port. This information is used to set up UDP communication between two
hosts that are both behind NAT routers. The protocol is defined in RFC 3489. STUN will usually work well
with non-symmetric NAT routers.
TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCP,
applications on networked hosts can create connections to one another, over which they can exchange
data or packets. The protocol guarantees reliable and in-order delivery of sender to receiver data.
TFTP Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a very
basic form of FTP; It uses UDP (port 69) as its transport protocol.
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UDP does not provide the reliability and ordering guarantees that TCP does; datagrams may arrive out of
order or go missing without notice. However, as a result, UDP is faster and more efficient for many
lightweight or time-sensitive purposes.
VAD Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing
wherein, the presence or absence of human speech is detected from the audio samples.
VoIP Voice over the Internet. VoIP encompasses many protocols. All the protocols do some form of
signaling of call capabilities and transport of voice data from one point to another. e.g.: SIP, H.323, etc.
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